Hi all
i need urgent help our entire switchboard is down only 5 days after it came
up.
this is the second time this has happened and i am thinking that asterisk is
not worth the trouble it gives.
mostly it runs without hassle. but around 2 weeks ago during the test phase
we rebooted the machine
Bristuff works fine with Beronet cards. As far as I can see there is no
difference between Beronet and Junghanns cards, not even physically.
As for chan_misdn, it is still in very early stages of development, so I
don't think you can expect features similar to those in bristuff, not to
mention it
Hi all,
I've a problem in DTMF dialog between * and a Cicpack Transit Node Class 4-5.
The call initiat by a mgcp phone pass by the cirpack and arrive in SIP on *,
everything is ok (negociation and phone call) but when we try to use the
voicemail, Asterisk don't understand DTMF.
Here are some l
On 2005.02.27 22:20 Jon Gabrielson wrote:
I have some extra FXS ports on my channel bank that I could
plug the modems into, but is asterisk's fax support good
enough for a production system? Everything I seem to read
seems to state that asterisk fax detection and fax send/receive
support is still
You are right Kevin, the dialplan continues if no agent is logged in,
anyway, I also had some typo on the next command after that that was
preventing it from cotinuing.
I must have missed the earlier reply you mentioned, sorry.
BTW, have seen the transfer problems some other user and I have talke
On Sun, 27 Feb 2005, Time Bandit wrote:
> On Sun, 27 Feb 2005 12:57:37 -0600, Anton Krall
> <[EMAIL PROTECTED]> wrote:
> > Guys.. which free softphone is the best,grandest,most recommended one out
> > there? based on your own experiences..
> Shameless plug : http://www.marccharbonneau.com/asteri
Anton Krall wrote:
I would like hear some advice from the guys using queues... seems that if no
agents are logged and a caller tries to enter a queue, asterisk just hangs
up on the caller... is there a way to define some sort of catchall so that
the caller can hear a message or maybe leave a messag
I have some extra FXS ports on my channel bank that I could
plug the modems into, but is asterisk's fax support good
enough for a production system? Everything I seem to read
seems to state that asterisk fax detection and fax send/receive
support is still very unreliable or is this only over long
Paul Fielding wrote:
I could be mistaken, but doesn't the license tie itself to the nics on
the server? I believe the Digium server will allow you to reregister as
much as you want as long as it's still got the same nics...
Paul
You do not need to re-register a key if the NICS have not changed.
On Mon, Feb 28, 2005 at 12:12:26AM +0200, BSDR wrote:
> I am trying to install a TE405P which is to be connected to E1 trunks.
> I have set the jumpers to E1 (closed), but modprobe wct4xxp still fails
> with the message: ZT_CHANCONFIG failed for channel 97 - which obviously
> means the card is st
[EMAIL PROTECTED] wrote:
Anybody have any issues running tdmoe on kernel 2.6+?
I've got Suse 9.1 + 9.2 running 2.6.5 and 2.6.8 respectively, and when I
enable dynamic spans between them, both boxes dump something similar to:
Badness in local_bh_enable at kernel/softirq.c:141
[] local_bh_enable+0x
How set the context of Transfer function?
There are 2 context in extensions.conf.
[con1]
exten => _0.,1,Dial(SIP/[EMAIL PROTECTED])
[con2]
exten => 812,1,Transfer(001345566);How
can use the dialplan of context con1?
Thanks!
Bill Chen
___
Asterisk-Use
How set the context of Transfer function?
There are 2 context in extensions.conf.
[con1]
exten => _0.,1,Dial(SIP/[EMAIL PROTECTED])
[con2]
exten => 812,1,Transfer(001345566);How
can use the dialplan of context con1?
Thanks!
Bill Chen
___
Asterisk-Use
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Greetings Everyone!
Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been
released. There is also a new tarball for Asterisk-sounds.
They are available for download on the digium FTP site:
ftp://ftp.asterisk.org/pub/asterisk/
ftp://ft
I'll look out for it, thanks!
Julius.
> Julius,
>
> I have just setup and installed phpconfig with the help of others on this
> mailing list. I didn't use CVS checkout as I don't have CVS installed.
>
> I am about to document the process for the Wiki which I hope will help :)
>
> C
>
> -Origi
hi all,
I have got a problem with asterisk "fromuser" field in sip.conf. Actually
I have got two asterisk servers communicating over sip. When a user from
Asterisk Server A calls a specific extension it is redirected to another
Asterisk Server B and that Asterisk Server B forwards it to a Soft
Im looking for a SIP one..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Domingo, 27 de Febrero de 2005 04:04 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandest Free Softphone
On Su
HELP NEEDED TURNING OFF THE cAPS lOCK KEY
:)
On Feb 25, 2005, at 20:07, Edward Banfa wrote:
Hello all,
Hi I would like to know how to configure a Mediatrix 1102 box to work
with my asterisk box. I have analog phones that i would like to connect
to my Mediatrix box and then connect the Mediatrix box
Im my case Im using Asterisk CVS-HEAD-02/20/05-00:42:04
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Domingo, 27 de Febrero de 2005 03:53 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
I am using canreinvite=no on all my sip.conf settings and also U use t on my
Dials... No luck so far.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M.
Sent: Domingo, 27 de Febrero de 2005 01:24 p.m.
To: Asterisk Users Mailing List - Non-Commer
I would like hear some advice from the guys using queues... seems that if no
agents are logged and a caller tries to enter a queue, asterisk just hangs
up on the caller... is there a way to define some sort of catchall so that
the caller can hear a message or maybe leave a message on a voicemail ?
Just check wether you have running mysql with the user name and host and
database provided in cdr_mysql.conf.
Thanks & Regards
Ritesh Jalan
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Monday, February 28, 2005 5:23 AM
Subject: [Asterisk-Users] CDR's are not stored in mysql
You misunderstand. Ofcourse I need to run the register program on the
machine itself. The point is I build them from images and every now and
then I roll out a new image. My question is, what do I need to preserve
from the previous image to keep the licences. Obviously reformatting
the disk and rer
On Mon, 2005-02-28 at 00:43 +0100, Ilija Poznic wrote:
> Hello my name is Ilija Poznic and I have a problem.
>
> My configuration is
> 1. Digium TDM4000P with one FXS.
> 2. AVM Fritz ISDN adapter (configured with capi).
>
> When I connect to my ISP and then start *. A
On Sun, 27 Feb 2005, Lee Howard wrote:
On 2005.02.27 11:28 Jon Gabrielson wrote:
You wouldn't happen to know how to do this would you?
I currently have a box with both hylafax and asterisk installed.
asterisk handles the dedicated voice lines over a t100p and
hylafax handles the dedicated fax l
I wonder if APP_RPT would be any help
see http://www.voip-info.org/wiki-Asterisk+cmd+Rpt and
http://zapatatelephony.org/app_rpt.html
Michael.
On Sun, 27 Feb 2005 05:06:27 -0500, Mark Phillips <[EMAIL PROTECTED]> wrote:
> I see that to be fraught with problems.
>
> Speaking as a radio ham,
Hi All
I seem to have a small problem with the music on
hold button on SJPhone.
I have 2 asterisk installations one from the Rapid
distribution and one from the latest CVS.
On the rapid dist when I press the music on hold
button on my SJPhone I get music on hold.
When I do the same I get
On February 27, 2005 08:59 pm, Andrew Kohlsmith wrote:
> On February 27, 2005 05:56 pm, BSDR wrote:
> > thanks for replying, here it the zaptel.conf:
>
> You need to close those four jumpers on the card that are clearly marked
> "J1-4 open for T1, closed for E1" or sommat.
I apologize, I did not s
On February 27, 2005 05:56 pm, BSDR wrote:
> thanks for replying, here it the zaptel.conf:
You need to close those four jumpers on the card that are clearly marked "J1-4
open for T1, closed for E1" or sommat.
-A.
___
Asterisk-Users mailing list
Asteris
Does any one know what happened with voice conduits? I have been trying to
reach them for nearly three weeks now. Their voice mail boxes are full and
writing email to them does not get any returns. Thoughts or sightings are
appreciated.
--
R.J.
__
On 2005.02.27 11:28 Jon Gabrielson wrote:
You wouldn't happen to know how to do this would you?
I currently have a box with both hylafax and asterisk installed.
asterisk handles the dedicated voice lines over a t100p and
hylafax handles the dedicated fax lines over a 4port serial card
with external
On 2005.02.27 10:36 Rich Adamson wrote:
Is it at all
realistic to assume that HylaFax could coexist with Asterisk on one
box?
Yes, you could have both Asterisk and HylaFAX running on the same box.
HylaFAX does not require a lot of resources (CPU, RAM, HDD) for small
deployments. I've had HylaFA
On 2005.02.27 09:30 Martijn van Oosterhout wrote:
On Sun, Feb 27, 2005 at 09:10:48AM -0800, Lee Howard wrote:
> Fax cannot handle a one-second delay. As Steve mentions in the
> article, per-spec fax has some timings (particularly silence in
> direction "switching") set at 75 ms +/- 20 ms. So if t
Have you checked the debug log? It will show you the actual mysql query used
and any possible errors.
-Matthew
> From: <[EMAIL PROTECTED]>
> Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List -
> Non-Commercial Discussion
> Date: Sun, 27 Feb 2005 20:53:17 -0300
> To:
> Subject: [Asteri
Hi guys,
I would llike to ask for your help on a problem I'm having with the cdr
functionality. I installed asterisk 1.0.4, and asterisk-addons-1.0.4 and
followed the procedures for installation and mysql configuration. Everything
seems fina. The cdr_mysql module is loaded, and I get no error
Robert Terzi wrote:
I found an old Dialogic card in an abandoned PC, that I think is a
Dialogic D/41EPCI based on some googling.. The lspci output says:
00:09.0 Bridge: PLX Technology, Inc. PCI <-> IOBus Bridge (rev 01)
Subsystem: Dialogic Corp: Unknown device 0529
I'm just getting starte
I have a recording studio in Brooklyn NY and I am seeking help in some
lessons and setting up. I have been learning linux and asterisk very
well thanks to lurking in the group and info on the voip-info. I would
like some one to come over and give me a tutorial, check my box out,
make sure every
at least 4 me.
Anyone knows what are the variables in an inbound IAX2 call who reflect the
actual codec and DNID, DNIS, original peer description, I'm only able to see
it during an iax debug
Timestamp: 3ms SCall: 1 DCall: 0 [66.98.146.34:5036]
VERSION : 2
CALLED NUM
Hello my name is Ilija Poznic and I have a problem.
My configuration is
1. Digium TDM4000P with one FXS.
2. AVM Fritz ISDN adapter (configured with capi).
When I connect to my ISP and then start *. Asterisks is registering me to SIP
provider iconnect. After that
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
> 1.0.5 is considered 1.0.x. Meaning, > v1.0.x means CVS-HEAD at the
> moment.
Thank you for explanation; now it is clear to me.
--
#Joseph
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/aster
On Sun, 2005-02-27 at 19:39 +, Kanishka Somaratne wrote:
> Hi
> how do i set an SIP users to make outgoing calls that is worth only
> $5. if they exceed $5 they can't make any calls. what i need is not a
> calling card, but to limit outgoing calls for SIP users depedning on
> a value i give.
>
Hi,
I guess I'd need to run Beronet quad and octo bri cards under bristuff to
get zaptel features (echo canceling, timing source) Am I right or could
I achieve this also with chan_misdn - their native driver ?
Running bristuff on Beronet cards is unsupported. Has anyone succesfully run
Berone
thanks for replying, here it the zaptel.conf:
span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109
loadzone= nl
defaultzone = nl
>hi,
>could you p
This is getting VERY annoying.
Is there anyone in here that has access to the list administration to
delete the user below???
>This is an automatically generated Delivery Status Notification.
>
>Delivery to the following recipients failed.
>
> [EMAIL PROTECTED]
>
>
>
>This is an automatic
Hi,
I am running Asterisk 1.0.5 Stable, and changing the pattern matching e.g to
10 digits made it call out instantly :-)
Not sure what your problem is.
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Duane
Sent: 27 February 2005 22:18
To: Asterisk Us
hi,
could you please provide your zaptel.conf, it's probably a configuration
problem.
BSDR a Ãcrit :
I am trying to install a TE405P which is to be connected to E1 trunks.
I have set the jumpers to E1 (closed), but modprobe wct4xxp still fails
with the message: ZT_CHANCONFIG failed for channel
Jeez, I need to work out the shortcut to send an email which I keep pressing
by accident!!
-Original Message-
From: C. Tomlinson [mailto:[EMAIL PROTECTED]
Sent: 27 February 2005 22:48
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] DISA and a lo
Hello,
Does anyone know of a web
based client that can be used with g723, g729 codec that may integrate with
Asterisk at all ?
I would consider commercial solutions as I understand g723 never
comes free
Thanks for any help
___
Guy,
I
think what Lyle meant was to put a wait as in dial -- wait ---
number.
Therefore the line is seized and then after a wait the number is
dialled.
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Guy C.
GuckenbergerSent: 27 Fe
Many thanks, that was the problem.
I didn't paste the context that forwards the call into the DISA context; it
had this in:
...DigitTimeout,5
..ResponseTimeout,10
Doh!
It works great with the mobile number, as I can pattern match 10 digits:
-Original Message-
From: [EMAIL PROTECTED]
[
> From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guy C.
Guckenberger
> Sent: Sunday, February 27, 2005 5:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Outbound call on TDM400P
>
>
> ok so I put the w
I just had problems getting through to this list so please accept this
test.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.d
(responder para [EMAIL PROTECTED], caso contrario não
receberei)
Boa tarde,
Gentileza estaremos fazendo manutençao do servidor
e esqueci que a lista é baseada em IP já que no seu arquivo de zonas do
asteriskbrasil.org listas.asteriskbrasil.org esta apontando para um IP,
para facil
ok so I put the wait in and still have the same
results.
Extensions.conf
exten => s,5,SetCallerID(${OUTCID})
exten => s,6,Wait(2) <-I
added this>
exten => s,7,Dial(${OUT}/${ARG1})
exten => s,8,Congestion
exten => s,107,Macro(outisbusy)
Im still only getting out ev
On Mon, February 28, 2005 8:33, Rod Bacon said:
> I agree. The following commands may also be of use...
Actually I disagree, I'm running 2 different asterisk servers, one with
1.0.5 and the other with CVS and I noticed this last night, the cvs
version attempts to send within a reasonable time, bu
I am trying to install a TE405P which is to be connected to E1 trunks.
I have set the jumpers to E1 (closed), but modprobe wct4xxp still fails
with the message: ZT_CHANCONFIG failed for channel 97 - which obviously
means the card is still in T1 mode. I have tried everything I can think
of, any id
Joseph wrote:
On Sun, 2005-02-27 at 15:53 -0600, Eric Wieling wrote:
Joseph wrote:
Note: I added all this section manually, when I compiled * 1.0.5 this
section wasn't there (I don't know why).
[featuremap]
;blindxfer => #1; Blind transfer
;disconnect => *0 ; Disconne
On Sun, 27 Feb 2005 12:57:37 -0600, Anton Krall
<[EMAIL PROTECTED]> wrote:
> Guys.. which free softphone is the best,grandest,most recommended one out
> there? based on your own experiences..
Shameless plug : http://www.marccharbonneau.com/asterisk/mediaxphone.php ;)
But it only works on Windows
On Sun, 2005-02-27 at 15:53 -0600, Eric Wieling wrote:
> Joseph wrote:
>
> > Note: I added all this section manually, when I compiled * 1.0.5 this
> > section wasn't there (I don't know why).
> >
> > [featuremap]
> > ;blindxfer => #1; Blind transfer
> > ;disconnect => *0
Hello,
The astGUiclient suite has it's own mailing list for questions like this:
https://lists.sourceforge.net/lists/listinfo/astguiclient-users
The easy fix is for you to set PHP globals to on and see if it works like
that first, also you could try making that directory writable.
MATT---
Joseph wrote:
Note: I added all this section manually, when I compiled * 1.0.5 this
section wasn't there (I don't know why).
[featuremap]
;blindxfer => #1; Blind transfer
;disconnect => *0 ; Disconnect
;automon => *1 ; One Touch Record
;atxfer =>
I've been playing with a variety of them over the past month.
The 'candidates' I've got down to are X-Lite, Firefly and SJphone. They all
have strengths and weaknesses, and all behave differently behind my firewall
(STUN client differences?). So far, I am happiest with the performance of
X-Lite.
On Sun, 2005-02-27 at 19:23 +, Julian J. M. wrote:
> You need to tell asterisk to stay in the media path. You must add 't'
> to the Dial options:
>
> exten => 21,1,Dial(${phone1},20,trwL(30:24:6))
>
> Or set canreinvite=no in your sip peer definition.
Still didn't work.
I had can
I agree. The following commands may also be of use...
.
.
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
.
.
- Original Message -
From: "Greg Hill" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List -
On Sun, 27 Feb 2005, C. Tomlinson wrote:
> I have just setup a DISA setup whereby people can dial in, authenticate, are
> given a dialtone and can then call out.
>
> Everything works however there is a 10 second delay after the user enters
> the number and presses #, until the system does anything
Hi,
I just tried your config setup out, seemed to work great.
I guess the reload scripts etc are a work in progress :p
I could edit files just fine though, threw the scrips in my /www dir and
didn't tweak anything else, I guess its all done due to my phpconfig
installation.
Drop us a line if yo
asterisk_on_oelf wrote:
Hi,
how can I see which codec a channel is currently using? I havn't found any
command to show this.
regards
Jens
sip show channels
iax2 show channels
--
Kristian Kielhofner
___
Asterisk-Users mailing list
Asterisk-Users@lists.digi
On Fri, Feb 25, 2005 at 09:15:19PM +0100, Martijn van Oosterhout wrote:
> You misunderstand. Ofcourse I need to run the register program on the
> machine itself. The point is I build them from images and every now and
> then I roll out a new image. My question is, what do I need to preserve
> from
On Mon, 2005-02-28 at 05:58, Riphagen, Ferdy wrote:
> Hello All,
>
> I have a macro and want to jump to another macro if a conditition is true or
> false.
> Asterisk is jumping to the next macro, but then the {ARG1} variable is not
> working anymore.
Try SetVar(SAVEARG=${ARG1}) in one macro then
Well, I thought I had my problem solved, but it is acting up again.
Hopefully this time I can provide enough information.
What I have is an * box setup with one X100P and TDM400 with one FXO and
one FXS. For my regular setup with interfacing with my PSTN and my
entire house with analog phones, the
Hi
how do i set an SIP users to make outgoing calls
that is worth only $5. if they exceed $5 they can't make any calls. what i need
is not a calling card, but to limit outgoing calls for SIP users depedning
on a value i give.
I use realtime asterisk.
Thank You
Kanishka
_
Hi,
how can I see which codec a channel is currently using? I havn't found any
command to show this.
regards
Jens
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or u
You wouldn't happen to know how to do this would you?
I currently have a box with both hylafax and asterisk installed.
asterisk handles the dedicated voice lines over a t100p and
hylafax handles the dedicated fax lines over a 4port serial card
with external modems.
It would be really nice if I cou
Yeah Im sure they are FXO. They came from Digium install in 2,3,4.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Lyman
Sent: Sunday, February 27, 2005 1:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Us
You need to tell asterisk to stay in the media path. You must add 't'
to the Dial options:
exten => 21,1,Dial(${phone1},20,trwL(30:24:6))
Or set canreinvite=no in your sip peer definition.
Julian J. M.
> but even adding it and commenting out "automon => *1" didn't work.
> and of cou
I guess you should do:
[macro-default]
exten => s,1,DBGet(do-not-disturb=DND/${ARG1})
exten => s,2,GotoIf($["${do-not-disturb}" = "YES"]?200)
...
exten => s,200,Macro(do_not_disturb,$ARG1) ; Call the macro, do not
jump directly like if it was a context
[macro-do_not_disturb]
exten => s,1,Wait(2)
Hello All,
I have a macro and want to jump to another macro if a conditition is true or
false.
Asterisk is jumping to the next macro, but then the {ARG1} variable is not
working anymore.
part of config:
[macro-default]
exten => s,1,DBGet(do-not-disturb=DND/${ARG1})
exten => s,2,GotoIf($["${do-no
Guys.. which free softphone is the best,grandest,most recommended one out
there? based on your own experiences..
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or upd
Guy C. Guckenberger wrote:
Ok all here is a strange one..
I have a TDM400P with 3 fxo modules. I can very rarely make an
outbound call to the PSTNabout once every 10 tries. However if I
use a analog phone pluged into the same phone line as one of the tdm
channels say channel 4, and I
Hello,
WavePad worked perfectly in the free version. Thank you.
l.
In data Sat, 26 Feb 2005 11:47:56 -0500, mattf <[EMAIL PROTECTED]>
ha scritto:
The free utility WavePad for Win32 will play and edit GSM files as well:
http://www.nch.com.au/wavepad/
To convert to/from GSM on Win32 you can use DB
> Any computer fax application that I know of will write the image to
> file, modern ones usually use TIFF. HylaFAX, mgetty+sendfax, efax,
> spandsp, etc. Then you just write the glue to make it deliver that fax
> image by e-mail. HylaFAX already has that built-in, and just requires
> some m
All the stuff on features.conf doesn't work for me too... I press *1 or # to
transfer a call and I get nothing...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Domingo, 27 de Febrero de 2005 12:21 p.m.
To: Asterisk Users Mailing List - No
I found an old Dialogic card in an abandoned PC, that I think is a
Dialogic D/41EPCI based on some googling.. The lspci output says:
00:09.0 Bridge: PLX Technology, Inc. PCI <-> IOBus Bridge (rev 01)
Subsystem: Dialogic Corp: Unknown device 0529
I'm just getting started with Asterisk to b
[snip]
> Try:
>
> exten => 21,1,Dial(${phone1},20,rw)
>
> Options should be combined.
>
> and
>
> exten => 21,1,Dial(${phone1},20,rwL(30:24:6))
>
> This limits the call to five minutes, warning every 60 seconds when four
> minutes are remaining. Keep in mind that time is speci
Put a 'w'ait in your dial command. * is
probably dialing too quickly after going off-hook.
Lyle
- Original Message -
From:
Guy C. Guckenberger
To: asterisk-users@lists.digium.com
Sent: Sunday, February 27, 2005 12:00
PM
Subject: [Asterisk-Users] Outbound c
Ok all here is a
strange one..
I have a TDM400P
with 3 fxo modules. I can very rarely make an outbound call to the
PSTNabout once every 10 tries. However if I use a analog phone pluged
into the same phone line as one of the tdm channels say channel 4, and I place
the analog phon
Hello All,
My test server has a dedicated public IP and was set up using the article
from ONLamp:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
except that I fixed the mailbox entries in the sip.conf.
Can some one please tell me what this means?
-
[EMAIL PROTECTED] wrote:
> I've added an introduction article about the ARA on my web
> site http://www.voip-forum.com/
>
> The same text is now also added to CVS head as
> README.realtime. On the same site, you will also find the
> news item about how we used
> Asterisk for a call from an airline
On Sun, Feb 27, 2005 at 09:10:48AM -0800, Lee Howard wrote:
> Fax cannot handle a one-second delay. As Steve mentions in the
> article, per-spec fax has some timings (particularly silence in
> direction "switching") set at 75 ms +/- 20 ms. So if the delay gets
> much larger than 75 ms, then th
Title: DISA and a long delay; ideas?
Hi,
I have just setup a DISA setup whereby people can dial in, authenticate,
are given a dialtone and can then call out.
Everything works however there
is a 10 second delay after the user enters the number and presses #, until the
system does anything.
[EMAIL PROTECTED] wrote:
> I didn't realize that the stable branch was never added to...
> So it will NEVER have any more features than it currently has???
1.0 STABLE will never have any more features.
1.2 STABLE will be released in the next 3 to 6 months, and it will
include all features that ha
On 2005.02.27 08:34 Martijn van Oosterhout wrote:
Hi,
I read it and found it very enlightening. I do have one question
regarding "Modems don't like relativity". It says modems need a
constant delay; is there a limit to what it can handle. For example,
would it be possible to configure a jitterbuffe
Title: DISA and a long delay; ideas?
Hi,
I have just setup a DISA setup whereby people can dial in, authenticate, are given a dialtone and can then call out.
Everything works however there is a 10 second delay after the user enters the number and presses #, until the system does anything.
On 2005.02.27 04:26 Rich Adamson wrote:
Back in the olden days, I recall several modem vendors bundling PC fax
software with their products. All of those old Win v3.1 apps created
a fax file (eg, pdf or otherwise) that could be distributed via email.
Well, I don't remember any of them doing PDFs.
Hello Steve.
It's an excellent read.
In two places you mention that V.34-Fax is 28,800 bps. Actually,
V.34-Fax has speeds ranging from 2400 baud to 33600 baud all using
V.34. And, while most V.34 connections are going to not probably be
more than 28,800 bps, I have seen sustained analog V.34 f
On Sun, Feb 27, 2005 at 04:14:46PM +0800, Steve Underwood wrote:
> Questions keep comming up about this, so I started writing something at
> http://www.soft-switch.org/foip.html . I think I covered the FAX over
> VoIP issues fairly completely. T.37 is pretty simple to explain. There
> is rather
I've added an introduction article about the ARA on my web site
http://www.voip-forum.com/
The same text is now also added to CVS head as README.realtime.
On the same site, you will also find the news item about how we used
Asterisk for a call from an airline jet above Greenland to Stockholm,
Swe
Hi, thanks for the reply,
I finally got asterisk to register my mediatrix, I am now able to dial
an analog phone (connected to the mediatrix, which in turn is connected
to *) from a soft phone (X-lite).I made a typo when creating the
corresponding user on my asterisk box, after i corrected that,
Title: [EMAIL PROTECTED]
It’s a pretty active forum too,
though if you have AMP related questions then sometimes better to ask them in
the AMP sourceforge forum
https://sourceforge.net/forum/?group_id=121515
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROT
1 - 100 of 137 matches
Mail list logo
